A New Optimum Jitter Protection for Conversational VoIP

Transcription

1 Proc. Int. Conf. Wirele Commun., Signal Proceing (Nanjing, China), 5 pp., Nov A New Optimum Jitter Protection for Converational VoIP Qipeng Gong, Peter Kabal Electrical & Computer Engineering, McGill Univerity Montreal,Quebec,Canada H3A 2A7 qi.gong@mail.mcgill.ca peter.kabal@mcgill.ca Abtract In Voice-over-IP, jitter buffer are introduced at both ide of the ender and the receiver to compenate for delay jitter. A longer buffer reduce the poibility of packet lo and packet diorder at the expene of increaing converational delay. In thi paper, we propoe a novel criterion for the calling quality of converational VoIP, including the effect of delay on interactivity of a converation. Uing thi criterion, we propoe a quality-baed playout cheduling algorithm with improved voice quality and reduced converational delay. The Simulation reult how that the propoed algorithm can achieve the bet calling quality compared with other algorithm. I. INTRODUCTION Voice over IP (VoIP) i a technology to exchange voice packet over the public Internet. The major challenge facing VoIP i how to match the voice quality with that provided by the traditional telephone network, i.e. PSTN. Therefore, for real-time voice communication over IP, the requirement on delay and packet lo are tringent to maintain proper QoS. However, the IP ervice model i bet effort, which make no guarantee on quality. To get higher voice quality in VoIP, effort have been made in the literature to reduce the delay, mooth the delay variation (alo known a jitter) and conceal packet loe [1]. Since the tranmiion delay for the delivered packet over IP are alway varying, the jitter-free playout cheme would lead to the increaing rate of packet lo and the miordered recovery voice, which impact the voice quality dramatically. In practice, a jitter buffer i introduced at the receiver ide to compenate for jitter effect. The ize of thi buffer can be fixed or adaptive. The trade-off of the ize of thi buffer i that the longer buffer increae the probability that a packet arrive before it i cheduled for playout, at the expene of increaing the converational delay to break the interactivity of the converation. The interactivity of the converation can be tranparent if the end-to-end delay i le than 150m [2]. ITU recommend that the upper limit of end-to-end delay i 400m [2]. Therefore, for application which experience the long network delay, it i deirable to keep the ize of jitter buffer a low a poible. Many olution have been propoed to deign a jitter buffer in the pat 25 year, and [3] gave a urvey and an analyi of mot claic approache in thi area. In current quality-baed buffer deign/optimization for VoIP, voice quality i ued a the key metric ince it link directly to end-uer perceived quality. Mot method (ee [3]) ue the E-model [4] to predict voice quality. Some algorithm, ee for example [5], ue packet lo rate (PLR) a the cot index for deigning a buffer ize. However, a low PLR doe not guarantee high quality, epecially in the cae that a PLR i reduced by chooing a large buffer to protect more late packet. For converational VoIP, a converational delay alo play a very important role for calling quality. A large converational delay would caue double talk, echo or even the termination of the converation. However, current E-model doe not take a converational delay into account. Therefore, we propoe a new criterion for the optimization of calling quality: maximize voice quality and reduce converational delay a much a poible. Accordingly, our quality-baed playout cheme i deigned by two part. To maximize voice quality, We ue the R factor in E-Model [4] a the cot index. A other qualitybaed algorithm, the etimate of delay ditribution i required in our cheme. Auming the hape of the ditribution tail i known, many work ue a priori elected ditribution to etimate the CDF of the delay ditribution, for example, Exponential in [6], Pareto in [7], Weibull in [8]. However, it wa noticed that the playout delay may be very enitive to the type of ditribution ued [9]. Therefore, we chooe the tatitical model baed on the hitogram which i more general and make no aumption on the delay ditribution. To reduce converational delay, we adopt the idea in [5] which take advantage of uing hangover frame, which trigger the compreion of the decoded voice to decreae a jitter buffer depth at the end of the talk-purt. The imulation reult how that our jitter buffer give better calling quality with horter converational delay. The contribution of thi paper are three-fold: 1) A new quality aement for converational VoIP which take into account both voice quality and converational delay 2) A new optimum jitter buffering baed on thi new criterion 3) A practical way to calculate converational delay which i more reaonable for human perception. The paper i organized a follow: the VoIP ytem ued

2 converational VoIP. A long converational delay would break up the interactivity of a converation. In thi paper, we alo give a practical way to calculate converational delay. talk-purt ilence Fig. 1: VoIP Sytem. in thi paper i introduced in Section II; in Section III, we propoe a new quality aement for converational VoIP which include voice quality and converational interactivity; an optimum playout chedule algorithm i propoed in Section IV. Finally, imulation and concluion are preented in Section V and Section VI. II. VOIP SYSTEM The VoIP ytem ued in thi paper i hown in Fig. 1. At the ender ide, the peech i firt encoded uing G711 encoder, and then encapulated into IP/UDP/RTP packet, whoe format follow [10]. In thi paper, the payload of a packet i 20m. At the receiver ide, the packet i firt put into the buffer to mooth out the delay jitter caued by variou delay over the network, and the packet inide the buffer are orted in order. The jitter buffer i deigned baed on optimization of calling quality which i decribed in Section IV. Each packet in the buffer i de-packetized into bit tream before it be decoded by the G.711 decoder. A packet lo concealment (PLC) technique i ued to handle the lot packet. Finally, the time-warping peech ignal uing PWSOLA i played out and the depth of jitter buffer i varied by (α 1) T F, where α i the caling factor and T F i the payload length of a packet. In Section IV, we will decribe how to deign α in detail. For converational application, both ide are ender and receiver. A jitter buffer i ued at both ide for the received packet. III. NOVEL QUALITY ASSESSMENT FOR CONVERSATIONAL VOIP For VoIP application, the calling quality i of the mot concern. With converational interactivity in mind, we propoe a new quality aement for converational VoIP, which take into account both voice quality and converational delay. The Mean Opinion Score (MOS), a cale from 1 to 5, i typically ued for aeing voice quality. However, uch a ubjective aement i cotly to get and i unuitable for a realtime application. Fortunately, ITU provide a computational aement baed on network component (delay and lo): the E-model, which can be ued to etimate MOS uing the formula in [4] (Equation B-4). Beide, a converational delay alo play an important role on perceived quality for A. E-Model E-Model i a computatioanl model provided by ITU-T which aee the combined effect of variation in everal parameter. The output of the model, R factor, can be ued to etimate cutomer opinion. According to [4], the R factor can be written a R =93.2 I e I d, (1) where I d i the delay impairment factor, and I e i the equipment impairment factor. I d can be derived by a implified fitting proce from [8], I d =0.024d +0.11(d 177.3) H(d 177.3), (2) where { 0, x < 0 H(x) = 1, x 0. The equipment impairment factor i codec dependent. For G.711 with PLC, it can be approximated a [9] I e = I ec + I ρ = ln(1 + 50ρ), (3) where I ec i the impairment caued by encoder, which i 0 for G.711, and ρ i the packet lo including network lo and the lo caued by jitter buffer. Defining I m a the impairment caued by lo and delay, it can be expreed a Then, (1) can be expreed a I m = I e + I d, (4) R =93.2 I m. (5) B. Converational Delay In [11], converational delay i defined by: the time interval between when Uer 1 top peaking and when Uer 1 hear the Uer 2 repone. Mathematically, we formulate it a: D cov = t play (i uer2 ) t end (j uer1 )+D codec (6) where D cov i the converational delay, D coded i the delay caued by the codec. t play ( ) i the time cheduled to play, t play ( ) i the ending time. i uer2 i Uer 2 firt packet of the firt talk-purt and j uer1 i Uer 1 lat packet of the lat talk-purt. Obviouly, to calculate D cov, the key i how to detect the tart and the end of a talk-purt. The firt packet of a talk-purt can be recognized by M field of RTP header, which i 1 for the firt voiced packet after ilence period [10]. The problem i how to define the end of the talk-purt. From our obervation on litening tet, we noticed that people tend to tart replying when the hangover packet are received. According to the definition of hangover in [12], the hangover packet i actually the unvoiced packet which i ent a a voiced packet to avoid peech clipping. So it i reaonable to define the end

3 of talk-purt when the firt hangover packet i perceived. In our ytem decribed in Section II, the latet verion of VAD/DTX from the G.729 [12] i ued on a receiver ide to detect the hangover packet to calculate D cov. Overall, our new quality aement can be expreed a Q converation = R + g(d cov ). (7) It i till an open quetion to define g( ), and it would be our future focu. Fortunately, the relation between Q converation and D cov i known: Q converation goe down when D cov goe up, and vice vera. Hence, Q converation can be optimized by maximizing R factor and minimizing D cov. IV. DESIGN OF PLAYOUT SCHEDULING ALGORITHM Human Speech conit of ilence and one or more. Packet loe during talk-purt decreae the perceived quality dramatically, while loe during ilence period caue almot no effect on the perceived quality. Therefore, mot playout cheduling algorithm tune a jitter buffer at the beginning of each talk-purt. Compared with continuouly updating approache, a per- approach take the advantage of producing a moother playout voice. In thi paper, we aim to deign a playout cheme baed on the optimization of calling quality. According to Section III, the optimization of calling quality for converational VoIP i equal to optimize voice quality, i.e. R factor, and keep the converational delay a low a poible. Recently, many quality-baed cheduling algorithm have been developed by maximizing the perceived voice quality, for example [7] and [8]. Although thee method can reduce a converational delay in ome ene, we believe that it can be done further. In [5], a playout algorithm wa propoed to reduce converational delay with zero initial buffering while providing jitter protection to mot voiced packet. Fig. 2 decribe the jitter management in detail. According to [5], a converational delay wa reduced by two tep: firt, playout the firt packet of a talk-purt a oon a it arrive; econd, compre the voiced packet in a jitter buffer whenever the hangover packet i detected. The propoed algorithm adopt thee two tep to reduce the converational delay. In [5], the maximum depth of the jitter buffer d max i predetermined and fixed for all talk-purt, the value i recommended to be more than 60m to achieve an acceptable level of packet lo. The problem i that the propagation delay ditribution i unknown, which make it hard to chooe a proper d max. Although a large value can be choen for d max to reduce the probability of packet eraure, thi increae the mouth-to-ear delay and degrade voice quality ( ee Fig. 3). Moreover, a large d max alo increae the converational delay and accordingly increae the rik of dirupting the converation interactivity. In Section V, the experimental reult would be preented to prove it. In thi paper, we propoe an optimum jitter buffering algorithm baed on the optimization of calling quality defined in (7), that i, the optimization of voice quality and keeping ender receiver jitter buffer depth (m) d1 d1 d1 t 1 t n 3 n 2 n 1 n t 5 L r t r r r r re 1 t 2 t 5 t n 3 t n t n n-3 n-2 n-1 n d2 d2 d1 d3 L t n 3 time t n Fig. 2: Delay-Free Jitter Buffer. the converational delay a low a poible. The following operation are performed: During a ilence period, comfort noie i played out every 20m, no matter whether the SID packet arrive or not. The jitter buffer ize i zero. Information about occurred packet loe and tranmiion delay are tored. When the firt voiced packet of the firt talk-purt arrive, PWSOLA i applied to tretch the decoded peech before it i played out. The jitter buffer ize increae by (α 1) T F. The d max i etimated baed on previouly tored information (window ize i 1000 packet) When the etimated d max i achieved, the decoded peech i not tretched any further. The depth of jitter buffer keep the maximum value d max and α =1. At the end of a converation turn, when the hangover i detected, PWSOLA i applied to compre the decoded peech before it i played out. The jitter buffer ize decreae by (1 α) T F. Compreion top when jitter depth i decreaed to zero. It i poible for hangover to happen in the middle of the talk-purt (ee Fig. 6), for example, the ilence gap within a word. In thi cae, we tretched the ubequent voiced packet a if it were the beginning of the talk-purt. A noticeable ilence gap can be avoided [5]. In our propoed algorithm, two variable, α and d max, need to be deigned. α i choen a follow: α 1+T p /T F (T p i one pitch period) during the tretching proce; α 1 T p /T F during the compreion proce; α =1during a ilence period or when the etimated d max i reached. For each talk-purt, d max i etimated baed on maximizing the expected voice quality. According to (5), the maximization of the R factor i equal to minimizing I m, which i the function of playout delay given the network lo. e t n

4 According to (2) and (3), (4) can be expreed a: I m = I d + I ρ =0.024d +0.11(d 177.3) H(d 177.3) +7ln ( (ρ n + ρ d ) ), with ρ n i the network lo and ρ d i the lo caued by buffer, which depend on the playout delay. ρ d can be calculated a ρ d =(1 ρ n )P (X >d)=(1 ρ n )(1 P (X d)) =(1 ρ n )(1 F (d)), in which F (d) i the CDF of delay. In thi paper, F (d) i calculated a a function of playout delay uing the hitogram of the mot recent w packet delay. In our imulation, w = 1000 packet. Fig. 3 how I m v. d for our trace. With the optimum playout delay d that minimize I m, it i eay to get an optimum d max for each talk-purt. For a converational application, thi deign i ued on the both ide of ender and receiver. Im Fig. 3: I m v. d. V. RESULTS To imulate the tranmiion over the internet, we ue a trace file from our databae to obtain the network delay for each packet, and Fig. 4 how the ditribution of the network delay. It i a typical one which preent mot feature of the trace in our databae for long ditance VoIP application. Fig. 5 how the PDF of the network delay. The network delay i half of the round-trip delay in the trace file, which wa collected in January, We ued hrping becaue of it ueful feature (timing the round trip delay in microecond and ending out packet every x milliecond). The data wa collected from the Internet connection between McGill Univerity (Canada) and Shanghai Jiao Tong Univerity (China), which i inide the China Education Network, ending packet every 20m for half an hour. The network lo i 1%, and min/avg/max of round-trip delay i 305m/308m/442m, and 153m/154m/221m for one-way delay accordingly. The converation ued in our imulation i from the recording of a real dialog, which conit of only two converation turn, in a ak-repone pattern. (8) (9) PDF Delay(m) Number of Packet x 10 4 Fig. 4: Network Delay (153m/154m/221m) Fig. 5: PDF of the Network Delay. Fig. 6 how the variable jitter buffering for one converation turn (the aking turn), which contain only one. The firt packet of talk-purt i tretched and played out immediately when it i received. The peech compreion begin when hangover i detected. In our experiment, the 81-t packet i the firt hangover packet. For the propoed algorithm, compreion tart at the 80-th packet, and for the cae of d max =60m, compreion tart at the 78-th packet, becaue the latter cae ha larger buffer and the 81-t packet i tored in buffer before the 78-th packet i played out. Although more packet are contracted, more buffer delay remain for the buffer with d max =60m, which increae the converational delay accordingly. Table I how the performance comparion of different jitter buffering algorithm: Optimum i the propoed algorithm; VJM 60m i the algorithm propoed in [5] with d max = 60m; VJM 80m i the algorithm propoed in [5] with d max =80m; Sun opt i the algorithm propoed in [8]. Even though the PLR i the highet, Optimum obtained the higher R factor (computed at the end of a converation turn) than VJM 60m and VJM 80m, becaue R factor i affected by not only the packet lo but alo the mouthto-ear delay. With the increae of d max, the packet lo decreae becaue more packet with large network delay can be played out, and thu decreae the negative effect on the

5 playout delay network delay Number of Packet tart tart end playout delay network delay end Number of Packet Fig. 6: Variable Jitter Protection: upper i the propoed buffering algorithm; bottom i the buffering algorithm in [5] with fixed depth of 60m R factor. However, the increment of d max alo increae the mouth-to-ear delay, and the negative effect on the R factor increae accordingly. In VJM 60m and VJM 80m, the impairment caued by mouth-to-ear delay i tronger than that caued by packet lo, wherea Optimum i deigned by the algorithm propoed in Section IV which concern both impairment on the R factor. The reult alo prove that lower PLR doe not guarantee a higher voice quality. Moreover, it can be een that Optimum achieve the lowet converational delay among all thee algorithm. According to the criteria of calling quality for converational VoIP in Section III, the propoed algorithm achieve the bet calling quality. VI. CONCLUSION In converational VoIP, cutomer care about the perceived calling quality the mot. For ome application, low converational delay i deired. In thi paper, we introduce a new quality aement including R factor and converational delay. And our optimum playout cheme i developed baed on optimizing thi calling quality, that i, maximizing voice quality etimated by E-model and keeping a converational delay a low a poible. Simulation reult how that the propoed algorithm can obtain the bet calling quality compared with other algorithm. REFERENCES [1] Y. J. Liang, N. Farber, and B. Girod, Adaptive Playout Scheduling uing Time-Scale Modification in Packet Voice Communication, in IEEE ICASSP 01, Salt Lake City, USA, Jan. 2001, pp [2] ITU, Recomendation G.114: One-way Tranmiion Time, ITU Std., May [3] L.Atzori and M. L. Lobina, Playout Buffering in IP Telephony: A Survey Dicuing Problem and Approache, IEEE Communication Survey & Tutorial, vol. 8, no. 3, pp , [4] ITU, The E-Model, a computational model for ue in tranmiion planning, ITU Std., Mar [5] M. Lee, J. McGowan, and M. C. Recchione, Enabling Wirele VoIP, Bell Lab Technical Journal, vol. 11, pp , Nov [6] M. Narbutt and L. Murphy, VoIP Playout Buffer Adjutment Uing Adaptive Etimation of Network Delay, in Proc. 18th International Teletraffic Congre (ITC-18), Berlin, Germany, 2003, pp [7] K. Fujimoto, S. Ata, and Murata, Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Application, Telecommunication Sytem, vol. 25, no. 3-4, pp , [8] L. Sun and E. Ifeachor, New Model for Perceived Voice Quality Prediction and their Application in Playout Buffer Optimization for VoIP Network, in Proc. IEEE ICC, Pari, France, Jun. 2004, pp [9] M. Ghanai and P. Kabal, Optimizing Voice-over-IP Speech Quality Uing Path Diverity, in Proc. IEEE Workhop on Multimedia Signal Proceing, Victoria,BC,Canada, Oct. 2006, pp [10] H. Schulzrinne, S. Caner, and V. Jacobon. (2003) RTP: A Tranport Protocol for Real-Time Application. [Online]. Available: [11] M. Yavuz, S. Diaz, R. Kapoor, M. Grob, P. Black, Y. Tokgoz, C. Lott, S. Guha, N. Dawani, and R.Jain, VoIP over CDMA2000 1xEV-DO Reviion A, IEEE Communication Magazine, vol. 44, pp , Feb [12] ITU, Recommendation G.729 Annex B: A Silence Compreion Scheme for G.729 Optimized for Terminal Conforming to Recommendation V.70, ITU Std., Jan TABLE I: Performance Comparion of Jitter Buffering Algorithm Buffering algorithm Condelay (m) R Factor Lo (%) Optimum VJM 60m VJM 80m Sun opt